webrtc::AudioProcessing Class Reference

#include <audio_processing.h>

Inheritance diagram for webrtc::AudioProcessing:

Classes
struct	AudioProcessingStatistics
struct	Config
struct	Stat
struct	Statistic

Public Types
enum	ChannelLayout {   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard,   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard }
enum	Error {   kNoError = 0, kUnspecifiedError = -1, kCreationFailedError = -2, kUnsupportedComponentError = -3,   kUnsupportedFunctionError = -4, kNullPointerError = -5, kBadParameterError = -6, kBadSampleRateError = -7,   kBadDataLengthError = -8, kBadNumberChannelsError = -9, kFileError = -10, kStreamParameterNotSetError = -11,   kNotEnabledError = -12, kBadStreamParameterWarning = -13, kNoError = 0, kUnspecifiedError = -1,   kCreationFailedError = -2, kUnsupportedComponentError = -3, kUnsupportedFunctionError = -4, kNullPointerError = -5,   kBadParameterError = -6, kBadSampleRateError = -7, kBadDataLengthError = -8, kBadNumberChannelsError = -9,   kFileError = -10, kStreamParameterNotSetError = -11, kNotEnabledError = -12, kBadStreamParameterWarning = -13 }
enum	NativeRate {   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000,   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000 }
enum	ChannelLayout {   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard,   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard }
enum	Error {   kNoError = 0, kUnspecifiedError = -1, kCreationFailedError = -2, kUnsupportedComponentError = -3,   kUnsupportedFunctionError = -4, kNullPointerError = -5, kBadParameterError = -6, kBadSampleRateError = -7,   kBadDataLengthError = -8, kBadNumberChannelsError = -9, kFileError = -10, kStreamParameterNotSetError = -11,   kNotEnabledError = -12, kBadStreamParameterWarning = -13, kNoError = 0, kUnspecifiedError = -1,   kCreationFailedError = -2, kUnsupportedComponentError = -3, kUnsupportedFunctionError = -4, kNullPointerError = -5,   kBadParameterError = -6, kBadSampleRateError = -7, kBadDataLengthError = -8, kBadNumberChannelsError = -9,   kFileError = -10, kStreamParameterNotSetError = -11, kNotEnabledError = -12, kBadStreamParameterWarning = -13 }
enum	NativeRate {   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000,   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000 }

Public Member Functions
virtual	~AudioProcessing ()
virtual int	Initialize ()=0
virtual int	Initialize (const ProcessingConfig &processing_config)=0
virtual int	Initialize (int capture_input_sample_rate_hz, int capture_output_sample_rate_hz, int render_sample_rate_hz, ChannelLayout capture_input_layout, ChannelLayout capture_output_layout, ChannelLayout render_input_layout)=0
virtual void	ApplyConfig (const Config &config)=0
virtual void	SetExtraOptions (const webrtc::Config &config)=0
virtual int	proc_sample_rate_hz () const =0
virtual int	proc_split_sample_rate_hz () const =0
virtual size_t	num_input_channels () const =0
virtual size_t	num_proc_channels () const =0
virtual size_t	num_output_channels () const =0
virtual size_t	num_reverse_channels () const =0
virtual void	set_output_will_be_muted (bool muted)=0
virtual int	ProcessStream (AudioFrame *frame)=0
virtual int	ProcessStream (const float *const *src, size_t samples_per_channel, int input_sample_rate_hz, ChannelLayout input_layout, int output_sample_rate_hz, ChannelLayout output_layout, float *const *dest)=0
virtual int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	ProcessReverseStream (AudioFrame *frame)=0
virtual int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0
virtual int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	set_stream_delay_ms (int delay)=0
virtual int	stream_delay_ms () const =0
virtual bool	was_stream_delay_set () const =0
virtual void	set_stream_key_pressed (bool key_pressed)=0
virtual void	set_delay_offset_ms (int offset)=0
virtual int	delay_offset_ms () const =0
virtual int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle)=0
virtual int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle)=0
virtual int	StopDebugRecording ()=0
virtual void	UpdateHistogramsOnCallEnd ()=0
virtual AudioProcessingStatistics	GetStatistics () const
virtual EchoCancellation *	echo_cancellation () const =0
virtual EchoControlMobile *	echo_control_mobile () const =0
virtual GainControl *	gain_control () const =0
virtual HighPassFilter *	high_pass_filter () const =0
virtual LevelEstimator *	level_estimator () const =0
virtual NoiseSuppression *	noise_suppression () const =0
virtual VoiceDetection *	voice_detection () const =0
virtual	~AudioProcessing ()
virtual int	Initialize ()=0
virtual int	Initialize (const ProcessingConfig &processing_config)=0
virtual int	Initialize (int capture_input_sample_rate_hz, int capture_output_sample_rate_hz, int render_sample_rate_hz, ChannelLayout capture_input_layout, ChannelLayout capture_output_layout, ChannelLayout render_input_layout)=0
virtual void	ApplyConfig (const Config &config)=0
virtual void	SetExtraOptions (const webrtc::Config &config)=0
virtual int	proc_sample_rate_hz () const =0
virtual int	proc_split_sample_rate_hz () const =0
virtual size_t	num_input_channels () const =0
virtual size_t	num_proc_channels () const =0
virtual size_t	num_output_channels () const =0
virtual size_t	num_reverse_channels () const =0
virtual void	set_output_will_be_muted (bool muted)=0
virtual int	ProcessStream (AudioFrame *frame)=0
virtual int	ProcessStream (const float *const *src, size_t samples_per_channel, int input_sample_rate_hz, ChannelLayout input_layout, int output_sample_rate_hz, ChannelLayout output_layout, float *const *dest)=0
virtual int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	ProcessReverseStream (AudioFrame *frame)=0
virtual int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0
virtual int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	set_stream_delay_ms (int delay)=0
virtual int	stream_delay_ms () const =0
virtual bool	was_stream_delay_set () const =0
virtual void	set_stream_key_pressed (bool key_pressed)=0
virtual void	set_delay_offset_ms (int offset)=0
virtual int	delay_offset_ms () const =0
virtual int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle)=0
virtual int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle)=0
virtual int	StopDebugRecording ()=0
virtual void	UpdateHistogramsOnCallEnd ()=0
virtual AudioProcessingStatistics	GetStatistics () const
virtual EchoCancellation *	echo_cancellation () const =0
virtual EchoControlMobile *	echo_control_mobile () const =0
virtual GainControl *	gain_control () const =0
virtual HighPassFilter *	high_pass_filter () const =0
virtual LevelEstimator *	level_estimator () const =0
virtual NoiseSuppression *	noise_suppression () const =0
virtual VoiceDetection *	voice_detection () const =0

Static Public Member Functions
static AudioProcessing *	Create ()
static AudioProcessing *	Create (const webrtc::Config &config)
static AudioProcessing *	Create (const webrtc::Config &config, NonlinearBeamformer *beamformer)
static AudioProcessing *	Create ()
static AudioProcessing *	Create (const webrtc::Config &config)
static AudioProcessing *	Create (const webrtc::Config &config, NonlinearBeamformer *beamformer)

Static Public Attributes
static const size_t	kMaxFilenameSize = 1024
static constexpr int	kNativeSampleRatesHz [4]
static constexpr size_t	kNumNativeSampleRates
static constexpr int	kMaxNativeSampleRateHz
static const int	kChunkSizeMs = 10

Member Enumeration Documentation

ChannelLayout [1/2]

enum webrtc::AudioProcessing::ChannelLayout

Enumerator
kMono
kStereo
kMonoAndKeyboard
kStereoAndKeyboard
kMono
kStereo
kMonoAndKeyboard
kStereoAndKeyboard

ChannelLayout [2/2]

enum webrtc::AudioProcessing::ChannelLayout

Enumerator
kMono
kStereo
kMonoAndKeyboard
kStereoAndKeyboard
kMono
kStereo
kMonoAndKeyboard
kStereoAndKeyboard

Error [1/2]

enum webrtc::AudioProcessing::Error

Enumerator
kNoError
kUnspecifiedError
kCreationFailedError
kUnsupportedComponentError
kUnsupportedFunctionError
kNullPointerError
kBadParameterError
kBadSampleRateError
kBadDataLengthError
kBadNumberChannelsError
kFileError
kStreamParameterNotSetError
kNotEnabledError
kBadStreamParameterWarning
kNoError
kUnspecifiedError
kCreationFailedError
kUnsupportedComponentError
kUnsupportedFunctionError
kNullPointerError
kBadParameterError
kBadSampleRateError
kBadDataLengthError
kBadNumberChannelsError
kFileError
kStreamParameterNotSetError
kNotEnabledError
kBadStreamParameterWarning

Error [2/2]

enum webrtc::AudioProcessing::Error

Enumerator
kNoError
kUnspecifiedError
kCreationFailedError
kUnsupportedComponentError
kUnsupportedFunctionError
kNullPointerError
kBadParameterError
kBadSampleRateError
kBadDataLengthError
kBadNumberChannelsError
kFileError
kStreamParameterNotSetError
kNotEnabledError
kBadStreamParameterWarning
kNoError
kUnspecifiedError
kCreationFailedError
kUnsupportedComponentError
kUnsupportedFunctionError
kNullPointerError
kBadParameterError
kBadSampleRateError
kBadDataLengthError
kBadNumberChannelsError
kFileError
kStreamParameterNotSetError
kNotEnabledError
kBadStreamParameterWarning

NativeRate [1/2]

enum webrtc::AudioProcessing::NativeRate

Enumerator
kSampleRate8kHz
kSampleRate16kHz
kSampleRate32kHz
kSampleRate48kHz
kSampleRate8kHz
kSampleRate16kHz
kSampleRate32kHz
kSampleRate48kHz

NativeRate [2/2]

enum webrtc::AudioProcessing::NativeRate

Enumerator
kSampleRate8kHz
kSampleRate16kHz
kSampleRate32kHz
kSampleRate48kHz
kSampleRate8kHz
kSampleRate16kHz
kSampleRate32kHz
kSampleRate48kHz

Constructor & Destructor Documentation

~AudioProcessing() [1/2]

virtual webrtc::AudioProcessing::~AudioProcessing ( )

inlinevirtual

~AudioProcessing() [2/2]

virtual webrtc::AudioProcessing::~AudioProcessing ( )

inlinevirtual

Member Function Documentation

AnalyzeReverseStream() [1/2]

virtual int webrtc::AudioProcessing::AnalyzeReverseStream ( const float *const *  data, size_t  samples_per_channel, int  sample_rate_hz, ChannelLayout  layout  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

AnalyzeReverseStream() [2/2]

virtual int webrtc::AudioProcessing::AnalyzeReverseStream ( const float *const *  data, size_t  samples_per_channel, int  sample_rate_hz, ChannelLayout  layout  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ApplyConfig() [1/2]

virtual void webrtc::AudioProcessing::ApplyConfig ( const Config &  config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ApplyConfig() [2/2]

virtual void webrtc::AudioProcessing::ApplyConfig ( const Config &  config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Create() [1/6]

static AudioProcessing* webrtc::AudioProcessing::Create ( )

static

Create() [2/6]

AudioProcessing * webrtc::AudioProcessing::Create ( )

static

Create() [3/6]

AudioProcessing * webrtc::AudioProcessing::Create ( const webrtc::Config &  config )

static

Create() [4/6]

static AudioProcessing* webrtc::AudioProcessing::Create ( const webrtc::Config &  config )

static

Create() [5/6]

static AudioProcessing* webrtc::AudioProcessing::Create ( const webrtc::Config &  config, NonlinearBeamformer *  beamformer  )

static

Create() [6/6]

AudioProcessing * webrtc::AudioProcessing::Create ( const webrtc::Config &  config, NonlinearBeamformer *  beamformer  )

static

delay_offset_ms() [1/2]

virtual int webrtc::AudioProcessing::delay_offset_ms ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

delay_offset_ms() [2/2]

virtual int webrtc::AudioProcessing::delay_offset_ms ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

echo_cancellation() [1/2]

virtual EchoCancellation* webrtc::AudioProcessing::echo_cancellation ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

echo_cancellation() [2/2]

virtual EchoCancellation* webrtc::AudioProcessing::echo_cancellation ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

echo_control_mobile() [1/2]

virtual EchoControlMobile* webrtc::AudioProcessing::echo_control_mobile ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

echo_control_mobile() [2/2]

virtual EchoControlMobile* webrtc::AudioProcessing::echo_control_mobile ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

gain_control() [1/2]

virtual GainControl* webrtc::AudioProcessing::gain_control ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

gain_control() [2/2]

virtual GainControl* webrtc::AudioProcessing::gain_control ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

GetStatistics() [1/2]

AudioProcessing::AudioProcessingStatistics webrtc::AudioProcessing::GetStatistics ( ) const

virtual

Reimplemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

GetStatistics() [2/2]

virtual AudioProcessingStatistics webrtc::AudioProcessing::GetStatistics ( ) const

virtual

Reimplemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

high_pass_filter() [1/2]

virtual HighPassFilter* webrtc::AudioProcessing::high_pass_filter ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

high_pass_filter() [2/2]

virtual HighPassFilter* webrtc::AudioProcessing::high_pass_filter ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [1/6]

virtual int webrtc::AudioProcessing::Initialize ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [2/6]

virtual int webrtc::AudioProcessing::Initialize ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [3/6]

virtual int webrtc::AudioProcessing::Initialize ( const ProcessingConfig &  processing_config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [4/6]

virtual int webrtc::AudioProcessing::Initialize ( const ProcessingConfig &  processing_config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [5/6]

virtual int webrtc::AudioProcessing::Initialize ( int  capture_input_sample_rate_hz, int  capture_output_sample_rate_hz, int  render_sample_rate_hz, ChannelLayout  capture_input_layout, ChannelLayout  capture_output_layout, ChannelLayout  render_input_layout  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Initialize() [6/6]

virtual int webrtc::AudioProcessing::Initialize ( int  capture_input_sample_rate_hz, int  capture_output_sample_rate_hz, int  render_sample_rate_hz, ChannelLayout  capture_input_layout, ChannelLayout  capture_output_layout, ChannelLayout  render_input_layout  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

level_estimator() [1/2]

virtual LevelEstimator* webrtc::AudioProcessing::level_estimator ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

level_estimator() [2/2]

virtual LevelEstimator* webrtc::AudioProcessing::level_estimator ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

noise_suppression() [1/2]

virtual NoiseSuppression* webrtc::AudioProcessing::noise_suppression ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

noise_suppression() [2/2]

virtual NoiseSuppression* webrtc::AudioProcessing::noise_suppression ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_input_channels() [1/2]

virtual size_t webrtc::AudioProcessing::num_input_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_input_channels() [2/2]

virtual size_t webrtc::AudioProcessing::num_input_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_output_channels() [1/2]

virtual size_t webrtc::AudioProcessing::num_output_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_output_channels() [2/2]

virtual size_t webrtc::AudioProcessing::num_output_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_proc_channels() [1/2]

virtual size_t webrtc::AudioProcessing::num_proc_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_proc_channels() [2/2]

virtual size_t webrtc::AudioProcessing::num_proc_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_reverse_channels() [1/2]

virtual size_t webrtc::AudioProcessing::num_reverse_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

num_reverse_channels() [2/2]

virtual size_t webrtc::AudioProcessing::num_reverse_channels ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

proc_sample_rate_hz() [1/2]

virtual int webrtc::AudioProcessing::proc_sample_rate_hz ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

proc_sample_rate_hz() [2/2]

virtual int webrtc::AudioProcessing::proc_sample_rate_hz ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

proc_split_sample_rate_hz() [1/2]

virtual int webrtc::AudioProcessing::proc_split_sample_rate_hz ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

proc_split_sample_rate_hz() [2/2]

virtual int webrtc::AudioProcessing::proc_split_sample_rate_hz ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessReverseStream() [1/4]

virtual int webrtc::AudioProcessing::ProcessReverseStream ( AudioFrame *  frame )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessReverseStream() [2/4]

virtual int webrtc::AudioProcessing::ProcessReverseStream ( AudioFrame *  frame )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessReverseStream() [3/4]

virtual int webrtc::AudioProcessing::ProcessReverseStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessReverseStream() [4/4]

virtual int webrtc::AudioProcessing::ProcessReverseStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [1/6]

virtual int webrtc::AudioProcessing::ProcessStream ( AudioFrame *  frame )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [2/6]

virtual int webrtc::AudioProcessing::ProcessStream ( AudioFrame *  frame )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [3/6]

virtual int webrtc::AudioProcessing::ProcessStream ( const float *const *  src, size_t  samples_per_channel, int  input_sample_rate_hz, ChannelLayout  input_layout, int  output_sample_rate_hz, ChannelLayout  output_layout, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [4/6]

virtual int webrtc::AudioProcessing::ProcessStream ( const float *const *  src, size_t  samples_per_channel, int  input_sample_rate_hz, ChannelLayout  input_layout, int  output_sample_rate_hz, ChannelLayout  output_layout, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [5/6]

virtual int webrtc::AudioProcessing::ProcessStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

ProcessStream() [6/6]

virtual int webrtc::AudioProcessing::ProcessStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_delay_offset_ms() [1/2]

virtual void webrtc::AudioProcessing::set_delay_offset_ms ( int  offset )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_delay_offset_ms() [2/2]

virtual void webrtc::AudioProcessing::set_delay_offset_ms ( int  offset )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_output_will_be_muted() [1/2]

virtual void webrtc::AudioProcessing::set_output_will_be_muted ( bool  muted )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_output_will_be_muted() [2/2]

virtual void webrtc::AudioProcessing::set_output_will_be_muted ( bool  muted )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_stream_delay_ms() [1/2]

virtual int webrtc::AudioProcessing::set_stream_delay_ms ( int  delay )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_stream_delay_ms() [2/2]

virtual int webrtc::AudioProcessing::set_stream_delay_ms ( int  delay )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_stream_key_pressed() [1/2]

virtual void webrtc::AudioProcessing::set_stream_key_pressed ( bool  key_pressed )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

set_stream_key_pressed() [2/2]

virtual void webrtc::AudioProcessing::set_stream_key_pressed ( bool  key_pressed )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

SetExtraOptions() [1/2]

virtual void webrtc::AudioProcessing::SetExtraOptions ( const webrtc::Config &  config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

SetExtraOptions() [2/2]

virtual void webrtc::AudioProcessing::SetExtraOptions ( const webrtc::Config &  config )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [1/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( const char  filename[kMaxFilenameSize], int64_t  max_log_size_bytes  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [2/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( const char  filename[kMaxFilenameSize], int64_t  max_log_size_bytes  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [3/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( FILE *  handle, int64_t  max_log_size_bytes  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [4/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( FILE *  handle, int64_t  max_log_size_bytes  )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [5/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( FILE *  handle )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecording() [6/6]

virtual int webrtc::AudioProcessing::StartDebugRecording ( FILE *  handle )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecordingForPlatformFile() [1/2]

virtual int webrtc::AudioProcessing::StartDebugRecordingForPlatformFile ( rtc::PlatformFile  handle )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StartDebugRecordingForPlatformFile() [2/2]

virtual int webrtc::AudioProcessing::StartDebugRecordingForPlatformFile ( rtc::PlatformFile  handle )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StopDebugRecording() [1/2]

virtual int webrtc::AudioProcessing::StopDebugRecording ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

StopDebugRecording() [2/2]

virtual int webrtc::AudioProcessing::StopDebugRecording ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

stream_delay_ms() [1/2]

virtual int webrtc::AudioProcessing::stream_delay_ms ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

stream_delay_ms() [2/2]

virtual int webrtc::AudioProcessing::stream_delay_ms ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

UpdateHistogramsOnCallEnd() [1/2]

virtual void webrtc::AudioProcessing::UpdateHistogramsOnCallEnd ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

UpdateHistogramsOnCallEnd() [2/2]

virtual void webrtc::AudioProcessing::UpdateHistogramsOnCallEnd ( )

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

voice_detection() [1/2]

virtual VoiceDetection* webrtc::AudioProcessing::voice_detection ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

voice_detection() [2/2]

virtual VoiceDetection* webrtc::AudioProcessing::voice_detection ( ) const

pure virtual

Implemented in webrtc::test::MockAudioProcessing, webrtc::test::MockAudioProcessing, webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

was_stream_delay_set() [1/2]

virtual bool webrtc::AudioProcessing::was_stream_delay_set ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

was_stream_delay_set() [2/2]

virtual bool webrtc::AudioProcessing::was_stream_delay_set ( ) const

pure virtual

Implemented in webrtc::AudioProcessingImpl, and webrtc::AudioProcessingImpl.

Member Data Documentation

kChunkSizeMs

static const int webrtc::AudioProcessing::kChunkSizeMs = 10

static

kMaxFilenameSize

static const size_t webrtc::AudioProcessing::kMaxFilenameSize = 1024

static

kMaxNativeSampleRateHz

static constexpr int webrtc::AudioProcessing::kMaxNativeSampleRateHz

static

Initial value:

=
      kNativeSampleRatesHz[kNumNativeSampleRates - 1]

kNativeSampleRatesHz

static constexpr int webrtc::AudioProcessing::kNativeSampleRatesHz

static

Initial value:

= {
      kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz}

kNumNativeSampleRates

static constexpr size_t webrtc::AudioProcessing::kNumNativeSampleRates

static

Initial value:

=
      arraysize(kNativeSampleRatesHz)

---

The documentation for this class was generated from the following files:

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_processing/include/audio_processing.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc